Re: [OpenSIPStack] TCP and TLS support

[<jo...@op...>]

Documentation is scarse for opensipstack, at least as of the moment.  
Please give more info on what type of application you would want to 
build so people can point you to the right direction.  For starters, 
CallSession and CallSessionManager subclass is the way to go for 
UA/Softphone implementation.


Marshall Jobe wrote:
> Yes...
>
> I want to access the SIP interface of opensipstack and let our code
> handle the RTP and voice (codecs and all).
>
> I just want to use the SIP portion. 
>
> I have found no technical documentation whatsoever and don't know where
> to go from here.
>
>
> -----Original Message-----
> From: jo...@op... [mailto:joe...@gm...] 
> Sent: Wednesday, October 22, 2008 8:49 PM
> To: ope...@li...
> Subject: Re: [OpenSIPStack] TCP and TLS support
>
> Marshall Jobe wrote:
>   
>> Help needed...
>> This is what we need to do.
>>
>> We have our own RTP stack that we want to use. We just want to use the
>> SIP
>> portion of the opensipstack and have the SIP portion tell us when to
>>     
> set
>   
>> up
>> an RTP call. We want to manage the RTP portion completely.
>>
>> How do we do this using opensipstack?
>>   
>>     
>
> Please clarify.  You want to use your own RTP stack.  Does this include 
> using your own codecs and audio input output steams as well?
>
>
>
>
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